We don't do calculations with different units (buffer offsets and bytes)
anymore but have functions for:
1) getting the number of bytes since the last discont
2) getting the offset (and pts/dts) at the last discont
and the previously added function to get the last offset and its distance from
the current adapter position.
https://bugzilla.gnome.org/show_bug.cgi?id=766647
API: gst_buffer_prev_offset
API: gst_buffer_get_offset_from_discont
The gst_buffer_get_offset_from_discont() method allows retrieving the current
offset based on the GST_BUFFER_OFFSET of the buffers that were pushed in.
The offset will be set initially by the GST_BUFFER_OFFSET of
DISCONT buffers, and then incremented by the sizes of the following
buffers.
The gst_buffer_prev_offset() method allows retrievent the previous
GST_BUFFER_OFFSET regardless of flags. It works in the same way as
the other gst_buffer_prev_*() methods.
https://bugzilla.gnome.org/show_bug.cgi?id=766647
The subclass should do that already, but just in case do it ourselves too as a
fallback. Without this, e.g. playbin will just wait forever if this fails
because it is triggered as part of an ASYNC state change.
Otherwise PTS and DTS will come out of sync if upstream continues to provide
PTS and not DTS, and we have to skip some data from the stream or PTS are not
exactly increasing with the duration of each packet.
https://bugzilla.gnome.org/show_bug.cgi?id=765260
gsttypefindhelper.c:485: Warning: GstBase: invalid "transfer" annotation for gsize: only valid for array, struct, union, boxed, object and interface types
If we don't store the value in prev_dts, we would over and over again
initialize the DTS from the last known upstream PTS. If upstream only provides
PTS every now and then, then this causes DTS to be rather static.
For example in adaptive streaming scenarios this means that all buffers in a
fragment will have exactly the same DTS while the PTS is properly updated. As
our queues are now preferring to do buffer fill level calculations on DTS,
this is causing huge problems there.
See https://bugzilla.gnome.org/show_bug.cgi?id=691481#c27 where this part of
the code was introduced.
https://bugzilla.gnome.org/show_bug.cgi?id=765096
It is calling do_sync(), which requires the STREAM_LOCK and PREROLL_LOCK to be
taken. The STREAM_LOCK is already taken in all callers, the PREROLL_LOCK not.
https://bugzilla.gnome.org/show_bug.cgi?id=764939
This is the best guess we can make if such a buffer reached the collect
pad. This is uncommon, we do expect parsers to have tried and fixed that
if possible (or needed).
https://bugzilla.gnome.org/show_bug.cgi?id=762207
Many parsers are storing tags only in pre_push_frame(), if we wouldn't check
afterwards we would push buffers before those tags and a lot of code assumes that
tags are available before preroll.
https://bugzilla.gnome.org/show_bug.cgi?id=763553
We have no guarantees about what flags are set on buffers we take
out of the GstAdapter. If we push out multiple buffers from the
first input buffer (which will have discont set), only the first
buffer we push out should be flagged as discont, not all of the
buffers produced from that first initial input buffer.
Fixes issue where the first few mp3 frames/seconds of data in push
mode were skipped or garbled in some cases, and the discont flags
would also trip up decoders which were getting drained/flushed for
every buffer. This was a regression introduced in 1.6 apparently.
Currently, the query values are being set even if the query itself was
determined to have failed. Fix this to ensure the values are only set in
case of a query success.
https://bugzilla.gnome.org/show_bug.cgi?id=760479
Watching videos with variant bitrate is common to have delta
more than 10 kbps, resulting in tag list spam.
Instead of relying on fixed 10 kpbs delta, it is better to
calculale the difference in percentage and update tag list
only when bitrate changes more than 2%.
https://bugzilla.gnome.org/show_bug.cgi?id=759055
This reverts commit 2c475a0355.
This causes issues with h264parse. It breaks timestamps as
there are headers in the middle of the stream and this patch
makes the timestamps for those differ from the ones that
are adjusted, creating a discontinuity and leading to sync
issues.
Otherwise the buffer was left with the original values and later would
be compared with other buffers that were converted to runninn time,
leading to bad interleaving of multiple streams.
https://bugzilla.gnome.org/show_bug.cgi?id=757961
baseparse tries to preserve timestamps from upstream if
it is running on a time segment and write that to
output buffers. It assumes the first DTS is going to be
segment.start and sets that to the first buffers. In case
the buffer is a header buffer, it had no timestamps and
will have only the DTS set due to this mechanism.
This patch prevents this by skipping this behavior for
header buffers.
https://bugzilla.gnome.org/show_bug.cgi?id=757961
They take a GstBaseSink instance as argument at not a GstPad. Rename the
argument to 'obj' which is not miss leading and in line with
GST_BASE_SINK_PAD(obj).
https://bugzilla.gnome.org/show_bug.cgi?id=756954
gst_segment_to_position might cause confusion, especially with the addition of
gst_segment_position_from_stream_time . Deprecated gst_segment_to_position
now, and replaced it with gst_segment_position_from_running_time.
Also added unit tests.
There exist cases where a reconfigure event was propagated from
downstream, but caps didn't change. In this case, we would
reconfigure only when the next buffer arrives. The problem is that
due to the allocation query being cached, the return query parameters
endup outdated.
In this patch we refactor the reconfigurating code into a function, and
along with reconfiguring when a new buffer comes in, we also reconfigure
when a query allocation arrives.
https://bugzilla.gnome.org/show_bug.cgi?id=753850
Explicitly keep track again whether upstream tags or parser tags
already contain bitrate information, and only force a tag update
for a bitrate if we are actually going to add the bitrate to the
taglist later. This fixes constant re-sending of the same taglist,
because upstream provided a bitrate already and we didn't add it,
so we didn't save the 'posted' bitrate, which would then in turn
again trigger the 'bitrate has changed too much, update tags'
code path. Fixes tag spam with m4a files for example.
https://bugzilla.gnome.org/show_bug.cgi?id=679768
In 0.10 there were no sticky events, and all tag events
sent would just be merged with the previously-received
tags. In 1.x we have sticky events, and the tags in the
tag event(s) should at all times carry the complete tags,
so we can't just push some tags and then just push tags
with just bitrates to update the bitrates, etc.
Instead we need to keep track of the upstream stream tags
received, of the tags set by the video decoder subclass,
and send an updated tag event with the combined tags
including our own bitrate tags (if applicable) whenever
the upstream tags, the subclass tags or any of our bitrates
change.
https://bugzilla.gnome.org/show_bug.cgi?id=679768
This is needed so that we can do proper tag handling
all around, and combine the upstream tags with the
tags set by the subclass and any extra tags the
base class may want to add.
API: gst_base_parse_merge_tags()
https://bugzilla.gnome.org/show_bug.cgi?id=679768
Use gst_pad_peer_query_duration() and remove a few
unnecessary levels of indentation. Rest of code might
looks a bit questionable, but leave it as is for now.
According to the design docs:
The ACCEPT_CAPS query is not required to work recursively, it can simply
return TRUE if a subsequent CAPS event with those caps would return
success.
So make it a shallow check instead of recursivelly check downstream.
https://bugzilla.gnome.org/show_bug.cgi?id=748635
GstPad has a flag for suggesting if the accept-caps
query should use intersect instead of the default
subset caps operation to verify if the caps would be
acceptable.
basetransform currently always uses the subset check and
this patch makes it honor the flag for using intersect
if it is set.
https://bugzilla.gnome.org/show_bug.cgi?id=748635
As of now, even for stream completly inside segment, there is no
guarantied that the DTS will be inside the segment. Specifically
for H.264 with B-Frames, the first few frames often have DTS that
are before the segment.
Instead of using the sync timestamp to clip out of segment buffer,
take the duration from the start/stop provided by the sub-class, and
check if the pts and pts_end is out of segment.
https://bugzilla.gnome.org/show_bug.cgi?id=752791
gst_query_find_allocation_meta() requires the query to be
writable to work. This patch ensure avoids taking a reference
on the query, so we can now check if a certain allocation meta
is present.
https://bugzilla.gnome.org/show_bug.cgi?id=752661
This line has no purpose, clearly gst_segment_do_seek() is doing
the right job, also, having the start time (a timestamp) be that
same as time (the stream time) is quite odd.
https://bugzilla.gnome.org/show_bug.cgi?id=750783
For files which are smaller than 1.5 seconds, the duration
estimation does not happen. So the duration will always be
displayed as 0. Updating the duration on EOS when the estimation
has not happened already
https://bugzilla.gnome.org/show_bug.cgi?id=750131
We must make the buffer writable to write its PTS and DTS, and also
reset its duration.
The behaviour is now the same as before commit c3bcbadd, except metas
might still be attached to the buffer extracted from the adapter.
https://bugzilla.gnome.org/show_bug.cgi?id=752092
This way we don't have to allocate/free temporary structs
for storing things in the queue array.
API: gst_queue_array_new_for_struct()
API: gst_queue_array_push_tail_struct()
API: gst_queue_array_peek_head_struct()
API: gst_queue_array_pop_head_struct()
API: gst_queue_array_drop_struct()
https://bugzilla.gnome.org/show_bug.cgi?id=750149
POOL meta just means that this specific instance of the meta is related to a
pool, a copy should be made when reasonable and the flag should just not be
set in the copy.
POOL meta just means that this specific instance of the meta is related to a
pool, a copy should be made when reasonable and the flag should just not be
set in the copy.
This preserves GstMeta properly unless the subclass does special things. It's
enough to make h264parse's stream-format/alignment conversion pass through
metas as needed.
https://bugzilla.gnome.org/show_bug.cgi?id=742385
All functions that return a GstBuffer or a list of them will now copy
all GstMeta from the input buffers except for meta with GST_META_FLAG_POOLED
flag or "memory" tag.
This is similar to the existing behaviour that the caller can't assume
anything about the buffer flags, timestamps or other metadata. And it's
also the same that gst_adapter_take_buffer_fast() did before, and what
gst_adapter_take_buffer() did if part of the first buffer or the complete
first buffer was requested.
https://bugzilla.gnome.org/show_bug.cgi?id=742385
The doc generator get confused with the inline structure. So
workaround by wrapping the inner of the structure with
public/private mark, and document that GST_COLLECT_PADS_DTS macro
shall be used to access this.
* Fix function name in sections.txt
* Add few missing or fix miss-named
* Workaround gtk-doc being confused with non typedef
types (loose track of public/private
There was few Since: mark missing their column. Also unify the way
we set the Since mark on enum value and structure members. These
sadly don't show up in the index.
These are not usable as they are, and can easily lead to crash
or leaks. This also silence warning from the scanner. If we manage to
make this usable, we can then remove that mark, it will require
to make this type boxed.
gstbasetransform.h:196: Warning: GstBase: "@submit_input_buffer" parameter unexpected at this location:
* @submit_input_buffer: Function which accepts a new input buffer and pre-processes it.
gstnetcontrolmessagemeta.c:103: Warning: GstNet: gst_buffer_add_net_control_message_meta: unknown parameter 'message' in documentation comment, should be 'addr'
Make gst_collect_pads_clip_running_time() function also store the
signed DTS in the CollectData. This signed DTS value can be used by
muxers to properly handle streams where DTS can be negative initially.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
Allow for sub-classes which want to collate incoming buffers or
split them into multiple output buffers by separating the input
buffer submission from output buffer generation and allowing
for looping of one of the phases depending on pull or push mode
operation.
https://bugzilla.gnome.org/show_bug.cgi?id=750033
In basesink functions gst_base_sink_chain_unlocked(), below code is used to
checking if buffer is late before doing prepare call to save some effort:
if (syncable && do_sync)
late =
gst_base_sink_is_too_late (basesink, obj, rstart, rstop,
GST_CLOCK_EARLY, 0, FALSE);
if (G_UNLIKELY (late))
goto dropped;
But this code has problem, it should calculate jitter based on current media
clock, rather than just passing 0. I found it will drop all the frames when
rewind in slow speed, such as -2X.
https://bugzilla.gnome.org/show_bug.cgi?id=749258
Since frame->priv->discont was cleared earlier,
GST_BASE_PARSE_FLAG_LOST_SYNC was never being set.
Take the chance to refactor the frame creation a bit to
organize the flags setting and reset.
https://bugzilla.gnome.org/show_bug.cgi?id=738237
Otherwise we're going to set a rather arbitrary DTS of segment.start (usually
0) for live sources, which confuses synchronization if the source started
capturing at a later time. And it's especially wrong for raw media, for which
we should not set any DTS at all.
https://bugzilla.gnome.org/show_bug.cgi?id=747731
It could be triggered by:
gst-launch-1.0 videotestsrc num-buffers=20 ! videcrop bottom=214748364 ! videoconvert ! autovideosink
Spotted while testing:
https://bugzilla.gnome.org/show_bug.cgi?id=743910
The flush-stop event should not restart the task for live sources unless
the element is playing. This was breaking seeks in pause with the rtpsrc.
https://bugzilla.gnome.org/show_bug.cgi?id=635701
Otherwise baseparse will consider empty streams to be an error while
an empty stream is a valid scenario. With this patch, errors would
only be emitted if the parser received data but wasn't able to
produce any output from it.
This change is only for push-mode operation as in pull mode an
empty file can be considered an error for the one driving the
pipeline
Includes a unit test for it
https://bugzilla.gnome.org/show_bug.cgi?id=733171
