Quite a few (broken?) files have a packet duration of 1ms, which is
most definitely wrong for either audio or video packets.
We therefore avoid using that value and instead use other metrics to
determine the buffer duration (like using the extended stream properties
average frame duration if present and valid).
Only error out when downstream returns:
* NOT_SUPPORTED
* ERROR
* NOT_NEGOTIATED
* NOT_LINKED
If we got _UNEXPECTED, we push an EOS downstream (since maybe only one
of the streams had gone EOS) and then stop the task silently.
In the case of WRONG_STATE we just need to stop silently
https://bugzilla.gnome.org/show_bug.cgi?id=600412
When on push mode and receiving an EOS event, asfdemux
should push all pending data because we might be dealing
with a broken file that has a preroll value higher
than its actual length.
Some (broken) streams don't have the extended stream properties in
the header, resulting in applying a duration of zero on outgoing
buffers.
Fixes#611473
Some files have payload with timestamps smaller than the preroll duration.
Instead of blindly substracting the preroll value (and ending up with
insanely high timestamps on the outgoing buffers), we make sure we
never go below 0.
Fixes#610432
We previously only aggregated flow returns after the while(push) loop,
which meant that in some cases we would end-up not properly aggregating
the flow returns.
This is based on the same flow aggregation algorithm as oggdemux.
Adds chained asfs handling to pull mode. It now checks if
there is a new asf header after the last packet (when it
is possible to know how many packets are) or it tries
checking if a processed packet that fails is an header
object.
Fixes#599718
Adds support for detecting and playing chained asfs
in push mode. asfdemux tries to detect a new asf start
by identifying the header object guid in a input buffer.
When it finds it, it resets its state, removing its pads
and creates new ones for the new file.
When receiving bogus data, we have to avoid subtracting a value
larger than 'size' from 'size' variable, resulting in a wrap
that would make 'size' a really large bogus value.
Fixes#599333
asf packets in rtp packets should come with their padding fields
set to 0 and the depayload must update them to the correct
value before pushing downstream
This also fixes a bug by which the first buffer (in a multi-packet mode)
passed to asf_demux_parse_packet() would have a GST_BUFFER_SIZE of the
full incoming buffer and not just of the single asf packet.
Fixes corrupted frames introduced by latest commit.
We now have a chance for packets to be collected before we send out the
newsegment. If we're not in accurate seeking (keyunit) it will set
the segment start/time to the keyframe's timestamp.
We now *always* seek to the keyframe just before our requested position.
When we encounter the first keyframe and we were not accurate (therefore doing
keyframe seeking), we update the segment start position to the keyframe timestamp.
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.
