This will still cause some timestamp jitter, but giving a hint as to the duration
rather than nothing seems to be a better idea.
Also, this allows some scenarios (like remuxing with asfmux) to estimate the total
duration using the accumulated packet duration (which will be correct).
The simple index entries also contain the number of packets one needs
to retrieve at a given position to get a full keyframe. We therefore
use that information to retrieve all those packets in one buffer when
working in pull-mode.
In gst_asf_demux_chain_headers, when 'goto wrong_type' was called
asfdemux tried to free a const pointer that had been cast to a
normal pointer variable.
We weren't taking the preroll into account previously, meaning that we
were always seeking preroll nanoseconds too early... resulting in a lot
of dropped packets (which are before the start time).
This brings quit a bit closer to as-fast-as-possible seeking in asf files.
Post global tags only after we've added our source pads, so that
tag events get sent downstream in addition to tag messages posted
on the bus. This makes sure tags can be picked up automatically
when transcoding, but also by tagreadbin/playbin2. Fixes#519721.
While we're at it, also add a container-format tag.
When we receive a DISCONT as input, don't clear our complete state but simply
mark a discont that will be put on the next buffer. The code will be able to
handle and throw away incomplete data.
Add some more debug info.
Remove an unused variable.
Don't overwrite the origin flow return by whatever flow we get
when trying to push the remaining internally queued payloads.
We want to do our eos logic, ie. send an EOS event or segment-done
message in any case. Makes things EOS properly when an EOS event
is forced upon the pipeline so that the source returns
FLOW_UNEXPECTED to a pulling asfdemux. Should fix#582056.
This also makes timestamps (more) consistent before and after a possible
seek, and moreover makes for reasonable position reporting in live stream
(whose payload timestamps should not be taken for granted).
* Improve newsegment handling, e.g. upstream might live in TIME.
* Only send newsegment if we have needed info.
* Avoid reading past end of data section.
The problem that happens is the following:
* A packet with multiple payloads comes in
* Those payloads get handled one by one
* The first payload contains the first audio payload with timestamp A
* The second payload contains the first video (key)frame with timestamp V (where V < A)
With the previous code, the following would happen:
* the first payload gets processed, then passed to queue_for_stream
* queue_for_stream detects it's the first valid timestamp received and stores
first_ts = A
* the second payload gets processed, then pass to queue_for_stream
* queue_for_stream detects the timestamp is lower than first_ts... and
discards it... resulting in losing the first keyframe of the video stream
We've been having this issue for *ages*... it's just that nobody noticed it
that much with playbin. But with playbin2's aggresive multiqueue handling, this
will result in multiqueue not being able to preroll (because the video decoder will
be dropping a ton of buffers before (maybe) receiving the next keyframe).
Tested with over 200 asf files, and they all play the first frame correctly now,
even the most braindead ones.
This might be caused by entering the if() line 1214 and then not having
any activated_streams.. resulting in reaching line 1267 without having
any valid flow value.
On win32, we're required to link to all the libraries used - including
ones only indirectly used by other libs. So, add gstaudio, gsttag, and
(for windows only) winsock.
Drop packets with an invalid replicated data length
instead of continuing with an invalid timestamp
and uninitialized payload metadata.
All other code assumes that the timestamps are valid.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_chain):
Remove duplicate and broken code for the streaming case and simply reuse
the much better working pull based code. Fixes#560348.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_add_video_stream):
Only copy sane aspect ratio values on the caps. Fixes#559682.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_loop):
Fix aggregated GST_FLOW_RETURN check for when to send an error message
on the bus.
Re-fixes #546859
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
Properly aggregate flow returns for both push and pull mode, so we shut
down if all pads are unlinked.
Fixes#546859.
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/dvdread/dvdreadsrc.c: (plugin_init):
* ext/lame/gstlame.c: (plugin_init):
* gst/asfdemux/gstasf.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
Original commit message from CVS:
* configure.ac:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_object):
Use correct error code for encrypted streams.
Original commit message from CVS:
* docs/plugins/gst-plugins-ugly-plugins-docs.sgml:
* docs/plugins/gst-plugins-ugly-plugins-sections.txt:
* ext/a52dec/gsta52dec.c:
* ext/amrnb/amrnbdec.c:
* ext/amrnb/amrnbenc.c:
* ext/amrnb/amrnbparse.c:
* ext/lame/gstlame.c:
* ext/mad/gstmad.c:
* ext/sidplay/gstsiddec.cc:
* gst/asfdemux/gstrtspwms.c:
* gst/mpegaudioparse/gstxingmux.c:
* gst/realmedia/rademux.c:
* gst/realmedia/rdtmanager.c:
* gst/realmedia/rtspreal.c:
* gst/synaesthesia/gstsynaesthesia.c:
Add missing elements to docs. Restore alphabetical order in section
file. Document mad (it was included in docs already).
Fix doc-markup: use convinience syntax for examples
(produces valid docbook), add several refsec2 when we have several
titles. Fix some types.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_stream_props):
Guard against division by 0 and fall back to 25/1 framerate.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_add_video_stream),
(gst_asf_demux_process_ext_stream_props):
Instead of adding a fixes 25/1 framerate to the video caps, use the
average frame duration in the extended properties of the video stream as
the framerate. Fixes#524346.
Original commit message from CVS:
Patch by:
Hans de Goede <j dot w dot r dot degoede at hhs dot nl>
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_seek_event):
If we don't have the position to seek to in our index first try
to convert from TIME to BYTES upstream and only if that fails
too use the old hack to simply seek to an earlier position
and let the sink drop everything before segment start.
Partially fixes bug #469930.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_ext_content_desc):
Convert tags that come as string into the type required by
GstTagList.
Original commit message from CVS:
* gst/asfdemux/Makefile.am:
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstrtspwms.c: (gst_rtsp_wms_before_send),
(gst_rtsp_wms_after_send), (gst_rtsp_wms_parse_sdp),
(gst_rtsp_wms_configure_stream), (_do_init),
(gst_rtsp_wms_base_init), (gst_rtsp_wms_class_init),
(gst_rtsp_wms_init), (gst_rtsp_wms_finalize),
(gst_rtsp_wms_change_state), (gst_rtsp_wms_extension_init):
* gst/asfdemux/gstrtspwms.h:
Move WMS RTSP extension from -good to here.
Port it to the new pluggable extension interface.
Original commit message from CVS:
* configure.ac:
* ext/mpeg2dec/gstmpeg2dec.c: (crop_buffer):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_descramble_buffer):
* gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcmdec_chain_raw):
Fix build against core CVS by not using deprecated API. Bump
requirements for new API (overdue anyway).
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_reset),
(gst_asf_demux_chain_headers),
(gst_asf_demux_parse_data_object_start), (all_streams_prerolled),
(gst_asf_demux_have_mutually_exclusive_active_stream),
(gst_asf_demux_check_activate_streams),
(gst_asf_demux_find_stream_with_complete_payload),
(gst_asf_demux_push_complete_payloads), (gst_asf_demux_loop),
(gst_asf_demux_activate_ext_props_streams),
(gst_asf_demux_process_object):
* gst/asfdemux/gstasfdemux.h:
Activate streams (ie. add the pads to the element) depending on
whether we actually get data for those streams within the ASF
preroll value specified. Currently only done in pull-mode though
(this will fix problems with playbin hanging on mms streams once
we use this in push-mode as well).
Original commit message from CVS:
* gst/asfdemux/asfpacket.c: (gst_asf_payload_queue_for_stream):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_reset),
(gst_asf_demux_init), (gst_asf_demux_push_complete_payloads),
(gst_asf_demux_process_file):
* gst/asfdemux/gstasfdemux.h:
Make all timestamps start from zero in pull-mode too; some small
clean-ups and FIXMEs here and there.
Original commit message from CVS:
* gst/asfdemux/asfpacket.c: (gst_asf_demux_parse_payload),
(gst_asf_demux_parse_packet):
If packet size is specified within the packet and smaller than
the actual packet size, don't parse beyond the size specified in
the packet (this makes us parse some cases of packets with single
compressed payloads cleanly, see e.g stream from #431318). Also
add a sanity check when parsing compressed single payloads.
Original commit message from CVS:
* gst/asfdemux/asfpacket.c: (gst_asf_payload_queue_for_stream):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_seek_index_lookup),
(gst_asf_demux_handle_seek_event),
(gst_asf_demux_push_complete_payloads):
Seeking improvements: honour the KEY_UNIT seek flag; after a seek, only
send data from the keyframe right before the new segment start to
make sure the decoder doesn't have to decode more than absolutely
necessary.
Original commit message from CVS:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_free_stream),
(gst_asf_demux_reset), (gst_asf_demux_parse_data_object_start),
(gst_asf_demux_loop), (gst_asf_demux_setup_pad),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_activate_stream),
(gst_asf_demux_parse_stream_object),
(gst_asf_demux_process_ext_stream_props),
(gst_asf_demux_process_queued_extended_stream_objects),
(gst_asf_demux_activate_ext_props_streams),
(gst_asf_demux_process_object):
* gst/asfdemux/gstasfdemux.h:
Refactor stream parse/activation a bit (stream activation heuristics
are still the same though); some more clean-ups.
Original commit message from CVS:
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init):
* gst/asfdemux/gstasfdemux.h:
Init debug category before using it.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_pull_data),
(gst_asf_demux_push_complete_payloads), (gst_asf_demux_loop):
Fix silly bug when we can't pull as much data as we want; don't
forget to announce pending tags in the new packet parsing code.
Original commit message from CVS:
* gst/asfdemux/Makefile.am:
* gst/asfdemux/asfpacket.c: (asf_packet_read_varlen_int),
(asf_packet_create_payload_buffer),
(asf_payload_find_previous_fragment),
(gst_asf_payload_queue_for_stream), (gst_asf_demux_parse_payload),
(gst_asf_demux_parse_packet):
* gst/asfdemux/asfpacket.h:
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_reset_stream_state_after_discont),
(gst_asf_demux_push_complete_payloads), (gst_asf_demux_loop),
(gst_asf_demux_setup_pad), (gst_asf_demux_descramble_buffer),
(gst_asf_demux_process_chunk):
* gst/asfdemux/gstasfdemux.h:
New packet parsing code: should put halfway decent timestamps on
buffers, and might even set the appropriate keyframe/discont buffer
flags from time to time (and even if it doesn't, I'm at least able
to debug this code); only used in pull-mode so far. Still needs
some more work, like payload extensions parsing and proper flow
aggregation, and stream activation based on preroll. Stay tuned.
Original commit message from CVS:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_free_stream),
(gst_asf_demux_seek_index_lookup),
(gst_asf_demux_handle_seek_event), (gst_asf_demux_get_stream),
(gst_asf_demux_setup_pad), (gst_asf_demux_add_video_stream),
(gst_asf_demux_process_ext_content_desc),
(gst_asf_demux_process_file), (gst_asf_demux_descramble_segment),
(gst_asf_demux_push_buffer), (gst_asf_demux_process_chunk),
(gst_asf_demux_process_segment), (gst_asf_demux_handle_data):
* gst/asfdemux/gstasfdemux.h:
Some clean-ups and small fixes: rename asf_stream_context structure to
AsfStream; inline some three-line utility functions that are only used
once anyway and get rid of their associated helper structs; make debug
category global so that it is used by the debug statements in the other
file as well; simplify gst_asf_demux_get_stream(); fix accidental
implicit initialisation of stream->last_buffer_timestamp to 0, which
would lead to missing timestamps on the first buffer; put fourcc format
into video caps to make certain proprietary wmv decoders happy (for the
case of WMVA in particular); play_time is offset by preroll as well, so
fix overreporting of duration for some files.
Original commit message from CVS:
* gst/asfdemux/asfheaders.c:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_free_stream),
(gst_asf_demux_reset), (gst_asf_demux_init),
(gst_asf_demux_activate), (gst_asf_demux_activate_push),
(gst_asf_demux_activate_pull), (gst_asf_demux_sink_event),
(gst_asf_demux_seek_index_lookup),
(gst_asf_demux_reset_stream_state_after_discont),
(gst_asf_demux_handle_seek_event),
(gst_asf_demux_handle_src_event), (gst_asf_demux_chain_headers),
(gst_asf_demux_chain), (gst_asf_demux_pull_data),
(gst_asf_demux_pull_indices),
(gst_asf_demux_parse_data_object_start),
(gst_asf_demux_pull_headers), (gst_asf_demux_loop),
(gst_asf_demux_setup_pad), (gst_asf_demux_process_file),
(gst_asf_demux_process_simple_index),
(gst_asf_demux_process_object),
(gst_asf_demux_send_event_unlocked), (gst_asf_demux_push_buffer),
(gst_asf_demux_handle_data), (gst_asf_demux_change_state):
* gst/asfdemux/gstasfdemux.h:
Make asfdemux work in pull mode where possible. If there's an index
at the end of the file, read it and use it for seeking purposes.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_file),
(gst_asf_demux_process_advanced_mutual_exclusion),
(gst_asf_demux_process_queued_extended_stream_objects),
(gst_asf_demux_process_object), (gst_asf_demux_change_state):
* gst/asfdemux/gstasfdemux.h:
Parse advanced mutual exclusion object and only add pads for
'hidden' streams (those in an extended stream header) that are
mutually exclusive with an already existing 'main stream' if
the broadcasting flag is not set. If the broadcasting flag is set,
assume that data for this stream isn't sent. (This should ideally be
solved better by making playbin more robust against this and/or by
making mmssrc send some information downstream about which streams
will be streamed). Fixes#353116.
Original commit message from CVS:
* gst/asfdemux/asfheaders.c:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_init),
(gst_asf_demux_sink_event), (gst_asf_demux_handle_seek_event),
(gst_asf_demux_identify_guid), (asf_demux_peek_object),
(gst_asf_demux_chain_headers), (gst_asf_demux_chain),
(gst_asf_demux_setup_pad), (gst_asf_demux_process_stream),
(gst_asf_demux_process_ext_content_desc),
(gst_asf_demux_get_object_header), (gst_asf_demux_process_header),
(gst_asf_demux_process_file), (gst_asf_demux_process_comment),
(gst_asf_demux_process_bitrate_props_object),
(gst_asf_demux_process_header_ext),
(gst_asf_demux_process_language_list),
(gst_asf_demux_process_ext_stream_props),
(gst_asf_demux_process_queued_extended_stream_objects),
(gst_asf_demux_process_object), (gst_asf_demux_change_state):
* gst/asfdemux/gstasfdemux.h:
Refactor and clean up header parsing and chain function a bit; get
rid of some cruft; make header parsing a tad more robust, fixing
#403188.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_sink_event):
Post an error if we receive an EOS event while still waiting for the
ASF header object to come through.
Original commit message from CVS:
Patch by: Xavier B. <xavierb gmail com>
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_get_guid),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_add_video_stream),
(gst_asf_demux_process_ext_content_desc),
(gst_asf_demux_process_data),
(gst_asf_demux_process_language_list),
(gst_asf_demux_process_ext_stream_props),
(gst_asf_demux_process_segment), (gst_asf_demux_handle_data):
Guard places where we assume that a certain amount of data is
available better against less data being available (should fix
infamous assertion crasher bug #336370). Also fixes a small
memory leak.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_get_gst_tag_from_tag_name),
(gst_asf_demux_process_ext_content_desc):
add a comment about a future change
* tests/check/elements/amrnbenc.c: (setup_amrnbenc),
(cleanup_amrnbenc):
* tests/check/elements/mpeg2dec.c: (setup_mpeg2dec),
(cleanup_mpeg2dec):
consistent pad (de)activation
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_seek_event),
(gst_asf_demux_process_data), (gst_asf_demux_process_file),
(gst_asf_demux_handle_src_query), (gst_asf_demux_change_state):
* gst/asfdemux/gstasfdemux.h:
Don't crash in the seek event handling code when playtime is 0,
as may be the case with live streams (#386218). Implement SEEKING
query so applications can query seekability without second-guessing
based on whether we have a duration or not.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_add_audio_stream):
The availability of extra codec data isn't something that
warrants debug messages at WARNING level (see #376958).
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_sink_event),
(gst_asf_demux_setup_pad), (gst_asf_demux_process_segment):
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_get_audio_stream),
(gst_dvd_demux_get_subpicture_stream):
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_get_video_stream),
(gst_mpeg_demux_get_audio_stream),
(gst_mpeg_demux_get_private_stream):
Active pads before adding them to the running element. Don't assert
on non-BYTE format newsegment events in asfdemux.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_get_gst_tag_from_tag_name),
(gst_asf_demux_process_ext_content_desc):
Erm, lets properly fix it. The only non-text tag that we support is
the track-number and that is an UINT. asfdemux was returning a GValue
initialized as INT. Further the Track and not the TrackNumber tag
(the latter is a string too).
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_get_gst_tag_from_tag_name),
(gst_asf_demux_process_ext_content_desc):
Skip tags that are unknown (was producing an uninialized GValue).
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_add_video_stream):
Use static pad templates with ANY caps for the source pads for
simplicity and to avoid warnings when creating pads for unhandled
codec IDs (#351795).
Original commit message from CVS:
* ext/dvdread/dvdreadsrc.c: (plugin_init):
* po/POTFILES.in:
Make custom error messages translatable.
* gst/asfdemux/gstasf.c: (plugin_init):
Remove setlocale() call, doesn't seem to be needed or recommended for
plugins, at least not according to gstreamer/docs/random/i18n.
Original commit message from CVS:
* gst/asfdemux/Makefile.am:
* gst/asfdemux/asfheaders.c:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_object):
Error out when the stream is encrypted (rather than feeding
garbage to the decoders). Fixes#349025.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_get_ext_stream_props_for_stream),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_change_state):
Find language codes for audio streams if they are available.
Original commit message from CVS:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_get_obj_stream),
(gst_asf_demux_process_stream),
(gst_asf_demux_process_language_list),
(gst_asf_demux_process_ext_stream_props),
(gst_asf_demux_process_queued_extended_stream_objects),
(gst_asf_demux_process_object), (gst_asf_demux_change_state):
* gst/asfdemux/gstasfdemux.h:
Parse extended stream properties objects and stream objects
hidden inside them (but delay creation of the appropriate
pads until after all the 'normal' stream objects have been
dealt with) (#343763). Also parse language list object.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
Skip chunks for unknown streams properly. Fixes broken sound
and/or video for files that have additional streams that
we don't recognise yet (e.g. if they are embedded in extended
stream properties). Partly fixes#343763.
Original commit message from CVS:
* gst/asfdemux/asfheaders.c:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_identify_guid),
(gst_asf_demux_process_header), (gst_asf_demux_push_obj),
(gst_asf_demux_pop_obj), (gst_asf_demux_process_object),
(gst_asf_demux_change_state):
* gst/asfdemux/gstasfdemux.h:
Add some more GUIDs and make debug log more readable
and easier to follow when parsing the headers.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_setup_pad),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_add_video_stream),
(gst_asf_demux_push_buffer):
* gst/asfdemux/gstasfdemux.h:
Handle unknown codec IDs/fourccs properly (#345879); send tag
events after newsegment event; fix use of GST_FOURCC_FORMAT
macro.
Original commit message from CVS:
* gst/asfdemux/gstasf.c: (plugin_init):
Call gst_riff_init() so the riff debug category gets set up
before it is being used.
Original commit message from CVS:
Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_sink_event),
(gst_asf_demux_push_buffer):
Send newsegment event only once per pad, fixes#336550.
Original commit message from CVS:
Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_setup_pad),
(gst_asf_demux_process_chunk):
* gst/asfdemux/gstasfdemux.h:
Subtract first timestamp from timestamps, so that
stream starts from 0; makes live streams that don't
start at 0 work again (fixes#317310, #336097).
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_data):
Read packet size, sequence and padsize in right order again
(fixes#332796; patch by: Fabrizio Gennari).
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_sink_event):
In sink event handler, release object lock again
_before_ sending EOS event downstream (#313838).
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_stream):
Do not error out on non-recognized streams. Ignore them and allow
playback of the other streams.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_file):
Take into account the file properties preroll value for
timestamping/newsegment. It's weird this value was commented out.
Original commit message from CVS:
2006-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
* ext/dvdnav/dvdnavsrc.c: (if):
* ext/dvdread/stream_labels.c:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_segment):
* gst/realmedia/rmdemux.c: (gst_rmdemux_loop):
fix up error domains, error strings, and use of translation
* po/POTFILES.in:
fix up this file, even though none of them are actually marked
for build yet.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_add_video_stream):
* gst/realmedia/rmdemux.c: (gst_rmdemux_chain),
(gst_rmdemux_add_stream), (gst_rmdemux_parse_mdpr):
Update for GST_FOURCC_FORMAT API change.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_init),
(gst_asf_demux_commit_taglist), (gst_asf_demux_process_comment),
(gst_asf_demux_process_ext_content_desc),
(gst_asf_demux_change_state), (gst_asf_demux_add_audio_stream),
(gst_asf_demux_add_video_stream), (gst_asf_demux_setup_pad):
* gst/asfdemux/gstasfdemux.h:
Improve metadata display, e.g. if the metadata comes before the
streams are loaded (which is perfectly valid).
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c: (gst_jpegdec_my_output_message),
(gst_jpegdec_my_emit_message), (gst_jpegdec_init):
Make jpegdec quiet on MJPEG decoding
* gst/asfdemux/README:
Fix mimetypes for MJPEG and H263
Original commit message from CVS:
* ext/dv/gstdvdec.c:
really fix bpp24/32 dvdec caps (classic rgba indeed)
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc):
don't send text tags if they are empty (bis repetita)
Original commit message from CVS:
* ext/dv/gstdvdec.c:
remove unneeded comment from dvdec
(related to DV 4CC codes in AVI files)
moved them in gstreamer/docs/random/mimetypes
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc):
don't send text tags if they are empty
fix mem leak on error path
* 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: (img_get_alpha_info):
* gst/ffmpegcolorspace/imgconvert_template.h:
adds BGR32 and BGRA32 to ffmpegcolorspace
(still bad colors, fixing it on next commit)
helps with dvdec outputing BGR32
Original commit message from CVS:
* ext/dv/demo-play.c: (main):
xvideosink -> xvimagesink
* ext/dv/gstdvdec.c:
change rgb 32/32 caps to 24/32 (no alpha)
change nb of channels to be a list (2 or 4, not 2)
change sample rate to be a list (32, 44.1, 48 kHz) not a range
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc):
Add 'date/year' to extracted metadata list
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc):
Extract TrackNumber metadata + clean up code
* gst/games/gstvideoimage.c: (gst_video_image_draw_rectangle):
Hope this is the good fix (var used unitialised)
Original commit message from CVS:
* configure.ac:
don't compile faad plugin if a RC of 2.0 is found
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc):
try to make Solaris compiler happier
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst/wavenc/riff.h:
Add AMR (VBR and CBR) ids to riff.h audio codec list
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc),
(gst_asf_demux_process_object):
Retrieve more tags from ASF files (Genre, AlbumTitle, Artist)
Original commit message from CVS:
* gst/asfdemux/README
* gst/wavenc/riff.h
* gst-libs/gst/riff/riff-ids.h
* gst-libs/gst/riff/riff-media.c
add new 4CC codes for h263 related codecs
fixes partially bug #155163
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_loop):
Align by packetsize, and assert that we a packet available before
playing. The first makes webstreams work (they often include
trailing padding data in a packet), the second allows pausing a
ASF stream in totem without getting demux errors afterwards.
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_handle_sink_event):
Set EOS on the element when processing an EOS event.
* ext/speex/gstspeexdec.h:
* ext/speex/gstspeexenc.h:
Only keep a const ptr to the mode
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data),
(gst_riff_create_audio_template_caps):
Allow WMAV3, with up to 6 channels.
* gst/asfdemux/gstasfmux.c: (gst_asfmux_request_new_pad):
Don't call gst_pad_set_event_function on a sink pad.
* gst/mpegstream/gstdvddemux.c:
(gst_dvd_demux_get_subpicture_stream),
(gst_dvd_demux_set_cur_audio), (gst_dvd_demux_set_cur_subpicture):
Copy the explicit caps that were set across to the cur_* pads,
instead of trying to use a possibly non-existent negotiated caps.
Reset the type of subpicture pads to UNKNOWN after calling init_stream,
so that the caps get set.
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
Don't touch buffer if it is of size 0 (fixes#151064).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_audio_caps_with_data):
Add codec_data handling (like asfdemux used to do).
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_add_video_stream):
Use riff-media for caps creation instead of our own (mostly
broken) copy of its functions.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (_read_var_length), (_read_guid),
(gst_asf_demux_process_segment), (gst_asf_demux_handle_data),
(gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event):
Prevent infinite loops. More correct error reporting.
* gst/auparse/gstauparse.c: (gst_auparse_chain):
Error out if negotiation fails.
* gst/playback/gstplaybasebin.c: (setup_source),
(gst_play_base_bin_change_state), (gst_play_base_bin_error),
(gst_play_base_bin_found_tag):
Error/tag forwarding. Pre-roll fixes for source errors on state
changes (e.g. "file does not exist") to prevent hangs.
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_check_caps_reset),
(gst_mad_change_state):
Allow for mp3 rate/channels changes. However, only very
conservatively. Reason that we *have* to enable this is smiply
because the mad find_sync() function is not good enough, it will
regularly sync on random data as valid frames and therefore make
us provide random caps as *final* caps of the stream. The best fix
I could think of is to simply require several of the same stream
changes in a row before we change caps.
The actual testcase that works now is #
* ext/ogg/Makefile.am:
* ext/ogg/gstogg.c: (plugin_init):
* ext/ogg/gstogmparse.c:
OGM support (video only for now; I need an audio sample file).
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_process_stream), (gst_asf_demux_video_caps),
(gst_asf_demux_add_video_stream):
WMV extradata.
* gst/playback/gstplaybasebin.c: (unknown_type):
Don't error out on single unknown-types after all. It's wrong.
If we found type of video and audio but not of a subtitle stream,
it will still error out (which is unwanted). Will find a better fix
later on.
* gst/typefind/gsttypefindfunctions.c: (ogmvideo_type_find),
(ogmaudio_type_find), (plugin_init):
OGM support.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_init),
(gst_asf_demux_loop), (gst_asf_demux_process_file),
(gst_asf_demux_process_data), (gst_asf_demux_handle_data),
(gst_asf_demux_process_object), (gst_asf_demux_get_stream),
(gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event),
(gst_asf_demux_handle_src_event), (gst_asf_demux_handle_src_query),
(gst_asf_demux_change_state):
* gst/asfdemux/gstasfdemux.h:
You know Chimaira? "I - HATE - EVERYTHING". Yeah, that's what this
feels like. I think we should set a new requirement for demuxers
from now on to implement sane loop functions, data loops, query
and seek functions before first commit into CVS. And this commit
fixes all of the above.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
- fix a mem leak and always propagate tags
- add WMV3 to known video codecs (but no decoder yet)
- replace "surplus data" at end of audio header for what
it is : codec specific data
- fix a typo
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c :
- freeing the event also frees its associated taglist,
that's unfortunate as we need taglist after that. fixed
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
don't write to memory we might not write to - g_convert does that
for us anyway
(gst_asf_demux_audio_caps):
conmment out gst_util_dump_mem
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
Fix some odd cases and fix BE metadata parsing of unicode16 text.
Original commit message from CVS:
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_get_type),
(gst_jpegenc_chain):
fix DURATION on outgoing buffers
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_sink_event):
debug using time formats
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_sink_link):
windows with width/height 0 generate X errors, so don't allow them
Original commit message from CVS:
* gst/asfdemux/Makefile.am:
* gst/asfdemux/asfheaders.c:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_get_type),
(gst_asf_demux_base_init), (gst_asf_demux_process_comment),
(gst_asf_demux_setup_pad):
* gst/asfdemux/gstasfdemux.h:
* gst/asfdemux/gstasfmux.c:
* gst/asfdemux/gstasfmux.h:
Add tagging support to demuxer, split out registration in its own
file instead of in demux (hacky), and prevent having some tables
in our memory multiple times (in asfheaders.h).
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_dispose):
actually free the URI string
* ext/mad/gstid3tag.c: (gst_id3_tag_src_event):
compute offset correctly when passing discont events
* ext/mad/gstid3tag.c: (gst_id3_tag_handle_event):
don't leak discont events
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps):
add some missing breaks so caps aren't copied randomly
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_get_video_stream):
if we realloc memory, we better use it
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps),
(gst_asf_demux_setup_pad):
Use 25fps as our "fake" fps value (marked for fixage in 0.9.x)
instead of 0. Reason is simple: some elements have a fps range
of 1-max instead of 0-max. So now ASF video actually works.
Original commit message from CVS:
* gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice):
Fix code that ignores return value of gst_buffer_merge().
(bug #114560)
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_descramble_segment):
* gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice): same
* testsuite/gst-lint: Check for above.
Original commit message from CVS:
2004-02-27 Benjamin Otte <otte@gnome.org>
* gst-libs/gst/audio/audio.h:
add macro to make sure header isn't included twice
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
don't use gst_buffer_free
* gst/playondemand/filter.func:
don't usae gst_data_free. Free data only once.
Original commit message from CVS:
2004-02-20 Andy Wingo <wingo@pobox.com>
* gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and
interleave respectively.
* gst/interleave/deinterleave.c: New plugin: deinterleave
(replaces on oneton).
* gst/interleave/interleave.c: New plugin: interleave.
* gst/interleave/plugin.h: Support file.
* gst/interleave/plugin.c: Support file.
* configure.ac: Remove intfloat and oneton, add interleave.
* ext/sndfile/gstsf.c: Handle events better.
* gst/audioconvert/gstaudioconvert.c: Change to support int2float
and float2int operation. int2float has scheduling problems as
noted in in2float_chain.
Original commit message from CVS:
2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_sink_event):
stop processing after EOS
Original commit message from CVS:
2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasfdemux.c:
* gst/asfdemux/gstasfmux.c: (gst_asfmux_put_guid),
(gst_asfmux_put_string), (gst_asfmux_put_wav_header),
(gst_asfmux_put_vid_header), (gst_asfmux_put_bmp_header):
lot's of fixes to make data extraction simpler and get the code
architecture and compiler independant. Add debugging category
* gst/goom/gstgoom.c: (gst_goom_change_state):
reset channel count on PAUSED=>READY, not READY=>PAUSED
Original commit message from CVS:
2004-01-15 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
Don't update the time of the clock
(gst_alsa_sink_loop):
sync to the clock given to alsasink, not the own clock
* sys/oss/gstosssink.c: (gst_osssink_chain):
sync to the clock
(gst_osssink_change_state):
activate the clock
* sys/ximage/ximagesink.c: (gst_ximagesink_chain):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain):
remove bogus code that made DISCONT events unhandled
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps):
explicitly case to double in _set_simple. (fixes 2nd warning in bug
#131502)
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_read_object_header),
(gst_asf_demux_handle_sink_event), (gst_asf_demux_audio_caps),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_video_caps):
convert g_warning because of wrong asf data to GST_WARNINGs (fixes
2nd warning in bug #131502)
Original commit message from CVS:
fixes for ASF:
- merge asfdemux and asfmux into one plugin
- make gstasf a plugin and not a lib (it accidently was one before)
Original commit message from CVS:
Sorry Dave... Add mpegversion=1 to mp3 caps everywhere so that the autoplugger uses mad and not faad for mp3 decoding. This should fix mp3 playback.
Original commit message from CVS:
Riff, EBML, fourcc etc. work. Not fully finished, but better than
what we used to have and definately worth a first broad testing.
I've revived rifflib. Rifflib used to be a bytestream-for-riff, which
just dup'ed bytestream. I've rewritten rifflib to be a modern riff-
chunk parser that uses bytestream fully, plus adds some extra functions
so that riff file parsing becomes extremely easy. It also contains some
small usability functions for strh/strf and metadata parsing. Note that
it doesn't use the new tagging yet, that's a TODO.
Avidemux has been rewritten to use this. I think we all agreed that
avidemux was pretty much a big mess, which is because it used all
sort of bytestream magic all around the place. It was just ugly.
This is a lot nicer, very complete and safe. I think this is far more
robust than what the old avidemux could ever have been. Of course, it
might contain bugs, please let me know.
EBML writing has also been implemented. This is useful for matroska.
I'm intending to modify avidemux (with a riffwriter) similarly. Maybe
I'll change wavparse/-enc too to use rifflib.
Lastly, several plugins have been modified to use rifflib's fourcc
parsing instead of their own. this puts fourcc parsing in one central
place, which should make it a lot simpler to add new fourccs. We might
want to move this to its own lib instead of rifflib.
Enjoy!
Original commit message from CVS:
Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files
Original commit message from CVS:
merge TYPEFIND branch. Major changes:
- totally reworked type(find) system
- all typefind functions are in gst/typefind now
- more typefind functions then before
- some plugins might fail to compile now because I don't have them installed and they
a) require bytestream or
b) haven't had their typefind fixed.
Please fix those plugins and put the typefind functions into gst/typefind if they don't have dependencies
Original commit message from CVS:
New typefind system:
* bytestream is now part of the core
* all plugins have been modified to use this new typefind system
* asf typefinding added
* mpeg video stream typefiding removed because it's broken
* duplicate typefind entries removed
* extra id3 typefinding added, because we've seen 4 types of files
(riff/wav, flac, vorbis, mp3) with id3 headers and each of these needs
to work. Instead, I've added an id3 element and let it redo typefiding
after the id3 header. this needs a hack because spider only typefinds
once. We can remove this hack once spider supports multiple typefinds.
* with all this, mp3 typefinding is semi-rewritten
* id3 typefinding in flac/vorbis is removed, it's no longer needed
* fixed spider and gst-typefind to use this, too.
* Other general cleanups
Original commit message from CVS:
Well, separated the stream from the RTP bits... RTP is disabled for now (will work on that long-term), and stream doesnt work yet, but it should be close now. Local playback works
Original commit message from CVS:
New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
Original commit message from CVS:
compatibility fix for new GST_DEBUG stuff.
Includes fixes for missing includes for config.h and unistd.h
I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately.
Original commit message from CVS:
Added code to handle split segments, changed src caps to video/avi to make
it work with ffmpeg. Correct time conversion code. Numerous minor bug fixes
and slight code cleanup.
Original commit message from CVS:
another batch of connect->link fixes
please let me know about issues
and please refrain of making them yourself, so that I don't spend double
the time resolving conflicts
Original commit message from CVS:
First step in giving us asf support is making this code widely available.
Now back to step 2 which used to be step 1 and get this code debugged so
it works :)
