Commit graph

1560 commits

Author SHA1 Message Date
Julien Moutte
0f80e462d9 gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
Original commit message from CVS:
2008-05-21  Julien Moutte  <julien@fluendo.com>

* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
instead of SOL_IP, works on more platforms.
* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
arguments.
2008-05-21 16:47:58 +00:00
Wim Taymans
2cdf18edff Some debug and comment fixes.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
Some debug and comment fixes.
* tests/examples/dynamic/addstream.c: (main):
Fix , to ;
2008-05-21 16:44:15 +00:00
Wim Taymans
c6b54c3d02 Don't use bad gst_element_get_pad().
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/decodetest.c: (new_decoded_pad_cb):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
(cleanup_decodebin):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(connect_element), (gst_decode_group_control_demuxer_pad):
* gst/playback/gstplaybasebin.c: (queue_remove_probe),
(queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
(mute_group_type):
* gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
(gst_play_bin_set_property), (handoff), (gen_video_element),
(gen_text_element), (gen_audio_element), (gen_vis_element),
(remove_sinks), (add_sink), (setup_sinks):
* gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
(gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
(gen_video_chain), (gen_text_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_request_pad):
* gst/playback/gsturidecodebin.c: (type_found), (setup_source):
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
(cb_newpad):
* gst/playback/test6.c: (new_decoded_pad_cb):
* tests/check/elements/audioconvert.c: (GST_START_TEST):
* tests/check/elements/audiorate.c: (test_injector_chain),
(do_perfect_stream_test):
* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/elements/gnomevfssink.c:
* tests/check/elements/textoverlay.c:
(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
* tests/check/pipelines/oggmux.c: (test_pipeline):
* tests/check/pipelines/streamheader.c: (GST_START_TEST):
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
* tests/examples/seek/scrubby.c: (make_wav_pipeline):
* tests/examples/seek/seek.c: (make_mod_pipeline),
(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
(make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline),
(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
(update_fill), (msg_buffering):
Don't use bad gst_element_get_pad().
2008-05-21 16:36:50 +00:00
Sebastian Dröge
736b181916 gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix copy & paste error in last commit.
2008-05-21 11:36:37 +00:00
Sebastian Dröge
7d605d4514 gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
other channel positions when source has SIDE channels and dest doesn't
or the other way around.
2008-05-21 11:30:58 +00:00
Henrik Eriksson
10ae17ced1 gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
Original commit message from CVS:
Patch by: Henrik Eriksson <henriken at axis dot com>
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
(gst_multi_fd_sink_get_property):
* gst/tcp/gstmultifdsink.h:
Add support for DSCP QOS. Fixes #469933.
2008-05-21 11:29:25 +00:00
Sebastian Dröge
d47bd6d7bc gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_normalize):
Prevent division by zero if the channel mix matrix contains only
zeroes.
2008-05-21 07:28:04 +00:00
Antoine Tremblay
a8dda35c1b gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
Close a buffer memory leak. Fixes bug #534071.
2008-05-21 06:45:22 +00:00
Sebastian Dröge
e66b0a6642 gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth so don't claim this on the pad tem...
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder also doesn't support audio/x-raw-int with width!=depth so don't
claim this on the pad template caps.
2008-05-21 05:48:05 +00:00
Sebastian Dröge
fcda3964dc gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Fix logic in last commit.
2008-05-20 12:26:32 +00:00
Sebastian Dröge
d76c4b4c65 gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Passthrough the channel positions if the number of output channels is
the same as the number of input channels, the input had a channel
layout and downstream requests no special one. We did this already for
> 2 channels but now it's also done for 1 channel. Fixes bug #533617.
2008-05-20 12:15:34 +00:00
Sebastian Dröge
b5a5d64713 gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Correctly set the default channel positions when converting to 8
channels.
2008-05-20 08:12:19 +00:00
Tim-Philipp Müller
7cb1276dac gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find):
Use data scan helper in aac typefinder and stop scanning
for headers when we've found a type. Also fix potential invalid
memory access when calculating the frame length.
2008-05-19 15:59:40 +00:00
Tim-Philipp Müller
cfc8f3c0d7 gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
(mpeg_sys_is_valid_pack):
Don't modify scan context when we return FALSE in ensure_data, so
it's possible to continue scanning, and we don't end up with a NULL
data pointer and a positive size, which might bite us the next time
we're called. Small constification.
2008-05-19 14:09:08 +00:00
Sebastian Dröge
05cf63634e gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder doesn't support 24 bit samples so don't claim it supports them
in the pad template caps.
2008-05-16 21:12:02 +00:00
Tim-Philipp Müller
d92ff26d29 gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Revert previous change which made basetransform handle buffer_alloc
and which breaks things badly in the non-passthrough case since it
returned buffers with a different (ie. sometimes smaller) size than
the size requested.
2008-05-14 13:57:41 +00:00
Sebastian Dröge
6720c5beb8 gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_class_init):
Correctly declare the supported endianness on the pad templates
and check for correct endianness in the set caps function. Adder
only supports native endianness.
Also use gst_element_class_set_details_simple().
2008-05-14 10:58:52 +00:00
Hannes Bistry
b9bc12afd8 gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
Original commit message from CVS:
Patch by: Hannes Bistry <hannesb at gmx dot de>
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversink.c:
(gst_tcp_server_sink_handle_server_read),
(gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
Fix regression in clientsrc because we did not add the fd to the poll
set anymore. Fixes #532364.
Do some cleanups here and there.
2008-05-13 16:02:19 +00:00
Sebastian Dröge
05349cc354 gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplay-marshal.list:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
Use correct marshallers. GstCaps are a boxed type and no GObject
subclass.
2008-05-13 13:04:24 +00:00
Sebastian Dröge
4d5870847f gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
Fix nv12<->nv21 conversion if stride is larger than width.
2008-05-13 09:14:44 +00:00
Tim-Philipp Müller
fed34307db gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Don't do lots of 4-byte peeks, but use the 'new' data scan helper
for this instead; don't check if we've found enough markers after
each and every step, it's enough to do that only if we've actually
found a new marker.
Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
2008-05-10 20:16:21 +00:00
Tim-Philipp Müller
104fed4d66 gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
(data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
(mpeg_video_stream_type_find):
Move scan helper thingy to the beginning of the file so we can use
it in other typefind functions. Rename it to something more
generic. Also improve handling of things towards the end of the
typefind data: peek as much as we can if we know the size of the
data, rather than just min_size.
2008-05-10 18:19:17 +00:00
Sebastian Dröge
531c6fb462 gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
Original commit message from CVS:
Based on a patch by:
Björn Benderius <bjoern dot benderius at axis dot com>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
* gst/ffmpegcolorspace/imgconvert_template.h:
Add conversions from/to NV12 and NV21 and conversions between those
two formats. Fixes bug #532166.
2008-05-09 08:34:52 +00:00
Edward Hervey
9fa3d7a294 gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Abort the h264 typefinding as soon as _peek() doesn't return anything,
which happens for example with files smaller than 128kb.
2008-05-08 17:35:44 +00:00
Sjoerd Simons
09163ca363 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
2008-05-08 06:20:42 +00:00
Sebastian Dröge
b9a285021c gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_transform_ip):
Return NOT_NEGOTIATED if we didn't set a process function yet for some
reason instead of crashing later. Might fix bug #509125.
2008-05-06 12:35:09 +00:00
Tim-Philipp Müller
fd54092a2a gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller  <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.
2008-05-06 12:12:16 +00:00
Wim Taymans
4a3db41f6d gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_pad):
When autoplugging fails, set the element back to NULL before
unreffing it.
2008-05-06 10:16:49 +00:00
Sebastian Dröge
9333eb4899 gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_sync),
(end_sami_element), (characters_sami):
Remove trailing, leading and double whitespaces.
Correctly timestamp buffers and output the last buffer too.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a simple unit test for SAMI parsing.
2008-05-05 12:33:05 +00:00
Young-Ho Cha
76e3ffb61c gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst/subparse/samiparse.c: (handle_start_sync),
(start_sami_element), (end_sami_element), (characters_sami),
(sami_context_reset):
Only output characters inside the "sync" elements. There could be
other elements like "style" that have some content but should
not be printed. Fixes bug #467911.
2008-05-05 11:14:48 +00:00
Sebastian Dröge
de277a5b2a gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (set_audio_mute),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(playbin_set_audio_mute):
Allow setting -1 as current-audio to mute the current audio stream,
similar to what is done for subtitles. Fixes bug #342294.
2008-05-05 10:03:51 +00:00
Sebastian Dröge
83f0729394 Remove some unused code.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
Remove some unused code.
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_free_noise_shaping):
Don't return before freeing the noise shaping history.
2008-05-04 15:02:20 +00:00
Tim-Philipp Müller
005c1c8636 gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (handle_buffer),
(gst_sub_parse_sink_event):
* gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
(tmplayer_parse_line):
Fix parsing of tmplayer subtitle variant where every single line contains
text and there isn't an empty line after each line to determine the
duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
making sure that we push out the last line of text without a duration if
there's still text left in the buffer at the end.
2008-05-03 15:45:23 +00:00
Tim-Philipp Müller
ee90cf1969 gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (feed_textbuf):
Fix detection of discontinuities based on the buffer offset (doesn't work
so well if no buffer offset is set) and also check for the DISCONT buffer
flag. This keeps the parser state from being reset after each buffer in
the unit test.
2008-05-03 15:39:04 +00:00
Tim-Philipp Müller
6de5983831 gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
Further fine-tuning: don't absolutely require sequence or GOP headers
(as introduced in the previous commit), but adjust the typefind
probabilities returned accordingly if we don't see them. Also make sure
picture header and first slice are somewhat close to each other (which
is not perfect but still better than requiring a fixed offset or having
no limit at all).
2008-05-03 12:09:16 +00:00
Stefan Kost
2b843ca69f gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/playback/test7.c:
Also include config.h when relying on defines from it. Fixes the
build. Its been a please to serve :)
2008-05-02 11:13:05 +00:00
Thijs Vermeir
32c304ad6f Add support for NV12 and NV21 in videotestsrc
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
(paint_setup_NV21), (paint_hline_NV12_NV21):
Add support for NV12 and NV21 in videotestsrc
2008-05-02 10:54:51 +00:00
Sebastian Dröge
abbce230e2 gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
* gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
(vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
(vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
(vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
(vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
(vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
(vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
(vs_image_scale_linear_RGB555):
Support 1x1 images as input and output as for example the BBC HQ new
streams have 1x1 GIFs in the playlists for some reason.
2008-05-02 10:02:05 +00:00
Tim-Philipp Müller
ea0d78e8e5 gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
(try_to_link_1):
If we can't activate one of the decoders we plugged in (such as,
say, musepackdec) for some reason (it might not support push mode,
for example), remove any pad probes that close_pad_link() might
have set up. This makes sure we later don't try to remove a probe
for a pad that doesn't exist any longer, and avoids nast warnings
and probably other things too.
2008-05-01 19:11:42 +00:00
Tim-Philipp Müller
f8977b9e9e gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
(plugin_init):
Rework mpeg video stream typefinding a bit more: make sure sequence,
GOP, picture and slice headers appear in the order they should and
that we've in fact at least had one of each; fix picture header
detection; decouple picture and slice header check - don't assume
they're at a fixed offset, there may be extra data in between. Also,
announce varying degrees of probability depending on what we found
exactly (multiple pictures, at least one picture, just sequence and
GOP headers). Finally, in _ensure_data(), take into account that we
might be typefinding smaller amounts of data, such as the first
buffer of a stream, so fall back to the minimum size needed as long
as that's available, instead of erroring out if there's less than
2kB of data. Fixes #526173. Conveniently also doesn't recognise the
fuzzed file from #399342 as valid.
2008-04-30 20:54:56 +00:00
Tim-Philipp Müller
5f6db60a4d gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
(mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
(mpeg_video_stream_type_find):
Refactor a bit: use context structure to track parsing offset and size of
available data and make the code a bit clearer. Fixes bad memory access
in #356937.
2008-04-30 14:37:52 +00:00
Michael Smith
802c45b10b gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/tcp/gstmultifdsink.c:
Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
is defined.
2008-04-28 22:18:49 +00:00
Stefan Kost
2b44c294ff gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
Original commit message from CVS:
* gst/playback/gstplaybin.c:
Remove obsolete streaminfo code and fix a leak. Fixes #529546
2008-04-24 08:19:35 +00:00
Sebastian Dröge
0c73cdcbc8 gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
(plugin_init):
Add "mpp" and "mp+" as possible extensions for MusePack files.
Add typefinding for MusePack StreamVersion 8 files and include the
stream version in the caps.
2008-04-19 20:06:59 +00:00
Edward Hervey
9e630c23db gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
Don't validate the payload if there isn't any.
Fixes #525915
2008-04-18 14:54:01 +00:00
Stefan Kost
e6528c39fe gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
Remove cpp style commented old code.
2008-04-15 19:06:00 +00:00
Stefan Kost
0d3c154ccf gst/playback/gstdecodebin2.c: Fix signal docs.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Fix signal docs.
2008-04-15 19:02:10 +00:00
Wim Taymans
f0738f6fd3 docs/design/draft-keyframe-force.txt: Fix typo.
Original commit message from CVS:
* docs/design/draft-keyframe-force.txt:
Fix typo.
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_handle_src_query):
Set buffering mode in the messages.
Set buffering percent in the query.
* tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
(do_stream_buffering), (do_download_buffering), (msg_buffering):
Do some more fancy things based on the buffering method in use.
2008-04-11 01:25:01 +00:00
Wim Taymans
e5bdd95038 gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
(gst_queue_src_checkgetrange_function):
Include extra buffering stats in the buffering message.
Implement BUFFERING query.
* gst/playback/gsturidecodebin.c: (do_async_start),
(do_async_done), (type_found), (setup_streaming), (setup_source),
(gst_uri_decode_bin_change_state):
Only add decodebin2 when the type is found in streaming mode.
Make uridecodebin async to PAUSED even when we don't have decodebin2
added yet.
2008-04-09 21:40:17 +00:00
Tim-Philipp Müller
7a29d716bd gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
2008-04-06 20:16:27 +00:00