Due to GstCollectPads sink pads list being not reliably
iteratable (when not inside the collected function) this
patch adds a sink pads list to qtmux to be used when iterating
sink pads on reset function.
Fixes#609055
Adds a new property to qtmux that sets a path to a file to write
and update data about the moov atom (that is not writen till the
end of the file). If the pipeline/app crashes during execution it
might be possible to recover the movie using the qtmoovrecover element.
qtmoovrecover is an element that is also a pipeline. It is not
meant to be used with other elements (it has no pads). It is merely
a tool/utilitary to recover unfinished qtmux files.
Fixes#601576
Following the previous qtmux commit, this patch tries
to use the new info added to the caps to fill the 'trak'
atom's fields and children atoms. This way qtmux will
use the late added 'codec_data' when h264parse adds
it in the following pipeline:
videotestsrc num-buffers=200 ! x264enc byte-stream=true ! \
h264parse output-format=0 ! qtmux ! \
filesink location=test.mov
Qtmux can accept caps renegotiation if the new caps is a
superset of the old one, meaning upstream added new info to
the caps. This patch still doesn't make qtmux update any
atoms info from the new info, but at least it doesn't
reject the new caps anymore.
A pipeline that reproduces this use case is:
videotestsrc num-buffers=200 ! x264enc byte-stream=true ! \
h264parse output-format=0 ! qtmux ! \
filesink location=test.mov
Reads the new caps added to qtdemux by commit
c917d65e6d
and adds its corresponding atoms.
Also adds support for image/x-jpc as it is the same
as image/x-jp2, except that the buffers need to be
boxed inside a jp2c isom box before muxing. To solve
this the QTPads now have a function that (if
not NULL) is called when a buffer is collected. This
function returns a replacement to the current collected
buffer.
Fixes#598916
Adds the mapping of 'classification' tags to writing of
'clsf' atoms for gppmux.
Based on a patch by: Lasse Laukkanen <ext-lasse.2.laukkanen@nokia.com>
Use the rounding version for improved sync between streams.
Small variations in the duration when muxing might lead to
cumullative wrong timestamping when demuxing.
Fixes#602936
Try to use timestamps even when the stream has out of order
timestamps, only fall back to durations when we detect an
out of order buffer. Improves sync between streams.
Adds support for muxing SVQ3 content. Usually this format
has decoder info that must be passed in the 'seqh' field
in the caps. It is also good to add the gama atom to make
quicktime not crash.
Fixes#587922
Do not wrongly add the result of the function to the
pointer to the buffer size. Instead, check the result
to see if the serialization was ok.
Based on a patch by: "Carsten Kroll <car@ximidi.com>"
Fixes#602106
When muxing streams, some can start later than others. qtmux
now handle this by adding an empty edts entry with the
duration of the 'lateness' to the stream's trak.
It tolerates a stream to be up to 0.1s late.
Fixes#586848
Keep track of the chunk durations to be able to add 3gr6
brand if it is a faststart file and the longest chunk is
smaller than a sec. Implemented according to 3gpp
TS 26.244 v6.4.0 (2005-09)
Fixes#584361
In faststart mode, there is no need to send the ftyp
right at the beginning of the stream. Waiting and sending it
only later (when the moov atom is ready to be sent) provides
us with more information about the stream and we can better
select the compatible brands.
Moves tags data initialization to the function that actually appends
the tags to the list. Fixes#582702
Also fixes some style caught by the pre-commit hook.
State change to READY and then back to PAUSED should still provide
the proper structures as are otherwise freshly available following
a request_new_pad.
Pointed out by Thiago Santos.
Original commit message from CVS:
* gst/quicktime/gstqtmuxmap.c: Add video/x-qt-part and video/x-m4-part
to caps so schroenc/schroparse can use it. Fixes#566958
Original commit message from CVS:
* gst/quicktime/gstqtmux.c: (gst_qt_mux_change_state):
Do not tempt or suggest to violate gst_collect_pads API specification.
Original commit message from CVS:
* ext/celt/gstceltenc.c:
* ext/celt/gstceltenc.h:
* ext/metadata/gstmetadatamux.c:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
Totally remove the internal taglists and fully use tagsetter. Fixes
various tag muxing issues.
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git
Even if we currently do not have a duration yet, assume seekable if
it looks like we'll likely be able to determine it later on
(which coincides with needed information to perform seeking).
Fixes#641047.
Even if VBR headers are missing, we can't guarantee that a stream is in
fact a CBR stream, so it's safer to let baseparse calculate the average
bitrate rather than assume a CBR stream. However, in order to make
/some/ metadata available before the requisite number of frames have
been parsed, this posts the bitrate from the non-VBR headers as the
nominal bitrate.
https://bugzilla.gnome.org/show_bug.cgi?id=641858
Rather than a fixed default frame count, estimate frame count to aim for
an interval duration depending on fps if available, otherwise use old
fixed default.
Also add a format flag to signal baseparse that subclass/format can provide
(parsed) timestamp rather than an estimated one. In particular, such "strong"
timestamp then allows to e.g. determine duration.
Don't unref the event if it hasn't been handled, because the caller
assumes it is still valid and might reuse it.
I ran into this problem when transcoding an AVI (with mp3 inside)
to gpp.
https://bugzilla.gnome.org/show_bug.cgi?id=639555
Fix copy'n'paste bug that made us look for the raw little endian
sync word twice instead of looking for the 14-bit LE sync word
as well. Fixes parsing of such streams (see #636234 for sample file).
That is, as such formats allow subclass to extract position from frame,
it is possible to extract duration (if not otherwise provided)
from (near) last frame, and a seek can fairly accurately target the required
position.
Fixes#631389.
Arrange for upstream as well as downstream flushing when seeking.
Also determine upstream size as well as seekability. Adjust some comments
to reality and employ debug statement in proper order.
Thanks to Felipe Contreras for the suggestion. This is partially
based on his patches and makes flacparse more than 3.5 times faster.
Looking for valid frame headers is unlikely to give false positives
because every frame header is at least 9 bytes long, contains a
14 bit sync code and a 8 bit checksum over the first 8 bytes.
Fixes bug #631200.
The first newsegment event will be send by the first call to
gst_base_parse_push_buffer() if necessary, posting the tags
before that is not a good idea. Instead do it from the
GstBaseParse::pre_push_buffer vfunc.
This reverts commit b5a3d60363.
Reverting this for now, since no one really seems to remember why this
property exists or what it could possibly be good for. It seems to have
been in the original mp3parse since the beginning of time and was back-
ported from there.
Seekability, like duration, etc is unlikely to change (frequently), and
the default assumption covers most cases, so let subclass set when needed.
At the same time, allow subclass to indicate if it has seek-metadata (table)
available, and possibly have it provide an average bitrate.
This allows the child class to chain its event handler with
GstBaseParse, so that subclasses don't have to duplicate all the default
event handling logic.
https://bugzilla.gnome.org/show_bug.cgi?id=622276
We wait to parse a minimum number of frames (10, arbitrarily) before
emiting bitrate tags so that our early estimates are not wildly
inaccurate for streams that start with a silence. If the stream ends
before that, we just emit the tags anyway.
While it _would_ be nicer to be specify the threshold to start pushing
the tags in terms of duration, this would introduce more complexity than
this merits.
https://bugzilla.gnome.org/show_bug.cgi?id=614991
This is optional because it's a quite expensive operation and it's very
unlikely that a non-frame is detected as frame after the header CRC check
and checking all bits for valid values. The overall frame checksums are
mainly useful to detect inconsistencies in the encoded payload.
When called from the GST_FLAC_PARSE_STATE_HEADERS case,
gst_flac_parse_hand_headers() does a gst_buffer_set_caps() on a buffer
with refcount > 1. This change handles this case by making the buffer
metadata_Writable.
https://bugzilla.gnome.org/show_bug.cgi?id=614037
This patch adds the get_frame_overhead() vfunc so that baseparse can
accurately calculate the min/avg/max bitrates for aacparse.
Note: The bitrate was being incorrectly calculated for ADTS streams
(it's not in the header as the code suggests).
This makes baseparse keep a running average of the stream bitrate, as
well as the minimum and maximum bitrates. Subclasses can override a
vfunc to make sure that per-frame overhead from the container is not
accounted for in the bitrate calculation.
We take care not to override the bitrate, minimum-bitrate, and
maximum-bitrate tags if they have been posted upstream. We also
rate-limit the emission of bitrate so that it is only triggered by a
change of >10 kbps.
Because config.h defines __MSVCRT_VERSION__, which should be defined
before inclusion of any system header.
Also fixes mpegdemux Makefile.am LIBADD typo.
Fixes#606665
Perform sanity check on type of seek, and only perform one that is
appropriately supported. Adjust downstream newsegment event
to first buffer timestamp that is sent downstream.
In particular, consider DISCONT == !sync, and allow subclass to query
sync state, as it may want to perform additional checks depending
on whether sync was achieved earlier on.
Also arrange for subclass to query whether leftover data is being drained.
In particular, (optionally) provide baseparse with a notion of frames per second
(and therefore also frame duration) and have it track frame and byte counts.
This way, subclass can provide baseparse with fps and have it provide default
buffer time metadata and conversions, though subclass can still install
callbacks to handle such itself.
After all, stream is as-is, and there is little molding to downstream's
taste that can be done. If subclass can and wants to do so, it can
still override as such.
Also handle the case gracefully where the subclass decides to drop
the first buffers and has no caps set yet. It's still required to
have valid caps set when the first buffer should be passed downstream.
In one case we extracted the sample rate index from the codec data
and saved it as sample rate rather than getting the real sample
rate from the table. Fix that, and also make sure we don't access
non-existant table entries by adding a small helper function that
guards against out-of-bounds access in case of invalid input data.
Create output caps from input caps, so we maintain any fields we
might get on the input caps, such as codec_data or rate and channels.
Set channels and rate on the output caps if we don't have input caps
or they don't contain such fields. We do this partly because we can,
but also because some muxers need this information. Tagreadbin will
also be happy about this.
Sending the flush-start event forward before taking the stream lock actually
works, in contrast to deadlocking in downstream preroll_wait (hunk 1).
After that we get the chain function being stuck in a busy loop. This is fixed
by updating the minimum frame size inside the synchronization loop because the
subclass asks for more data in this way (hunk 2).
Finally, this leads to a very probable crash because the subclass can find a
valid frame with a size greater than the currently available data in the
adapter. This makes the subsequent gst_adapter_take_buffer call return NULL,
which is not expected (hunk 3).
The problem is that after a discont, set_min_frame_size(1024) is called when
detect_stream returns FALSE. However, detect_stream calls check_adts_frame
which sets the frame size on its own to something larger than 1024. This is the
same situation as in the beginning, so the base class ends up calling
check_valid_frame in an endless loop.
Baseparse internaly breaks the semantics of a _chain function by calling it with
buffer==NULL. The reson I belived it was okay to remove it was that there is
also an unchecked access to buffer later in _chain. Actually that code is wrong,
as it most probably wants to set discont on the outgoing buffer.
1) We need to lock and get a strong ref to the parent, if still there.
2) If it has gone away, we need to handle that gracefully.
This is necessary in order to safely modify a running pipeline. Has been
observed when a streaming thread is doing a buffer_alloc() while an
application thread sends an event on a pad further downstream, and from
within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
while the streaming thread has its buffer_alloc() in progress.
If we get GAP samples, there is no need to transmitt it.
In some situations, microphone is muted, we can drop net traffick
usage to ~1 kbit/s. Without patch it will stay ~20 kbit/s
Parse session control attributes when no media control attribute is
present. Threat * control attributes as an empty string, just like the
spec says.
Fixes#646800
... by forcing a state changed to PLAYING, which should otherwise be a
no-op as elements should already be in that state.
In particular, jitterbuffer needs new base_time as soon as possible to perform
proper timing (e.g. eos timeout handling) and can't wait for the new base_time
that will be distributed when the whole pipeline returns to PLAYING.
See bug #646397.
This option allows the videomixer2 element to output a valid alpha
channel when the inputs contain a valid alpha channel. This allows
mixing to occur in multiple stages serially.
The following pipeline shows an example of such a pipeline:
gst-launch videotestsrc background-color=0x000000 pattern=ball ! video/x-raw-yuv,format=\(fourcc\)AYUV ! videomixer2 background=transparent name=mix1 ! videomixer2 name=mix2 ! ffmpegcolorspace ! autovideosink videotestsrc ! video/x-raw-yuv,format=\(fourcc\)AYUV ! mix2.
The first videotestsrc in this pipeline creates a moving ball on a
transparent background. It is then passed to the first videomixer2.
Previously, this videomixer2 would have forced the alpha channel to
1.0 and given a background of checker, black, or white to the
stream. With this patch, however, you can now specify the background
as transparent, and the alpha channel of the input will be
preserved. This allows for further mixing downstream, as is shown in
the above pipeline where the a second videomixer2 is used to mix in a
background of an smpte videotestsrc. So the result is a ball hovering
over the smpte test source. This could, of course, have been
accomplished with a single mixer element, but staged mixing is useful
when it is not convenient to mix all video at once (e.g. a pipeline
where a foreground and background bin exist and are mixed at the final
output, but the foreground bin needs an internal mixer to create
transitions between clips).
Fixes bug #639994.
Previously the chain function was working sample frame based. In each cycle it
was checking if it is time to run a fft or if it is time to send a message.
Now we changed the data transform functions to work on a block of data and
calculate the max length until either {end-of-data, do-fft, do-msg}. This allows
us also to avoid the duplicated code for the single and multi-channel case (as
the transformers have the same signature now).
Even though we wrap around the accumulated second, we still need to add the
error in the same cycle. Increase the todo in the same conditional as afterwards
the accumulated error will be below one second.
AUTHOR only existed in an old version of the spec and ARTIST is
the new replacement for this. We are still reading both to still
be compatible with old files.
Fixes bug #644875.
Before it was possible that we run an extra fft when the time for sending a new
message is due. Only do this if we have not run the fft for the interval at all.
Don't check the format for each sample frame to read. We can make that decission
in _setup already. This is still not ideal as we call the function per frame.
Ideally we determine how many samples we can copy and have a loop in the input
reader. As an alternative we might also consider to use the fft variants for the
various formats and not convert to float for all cases - we would still need to
mix or deinterleave though.
In case server-side fails to perform seek, i.e. PLAY at non-zero requested
position, recovery so far would arrange for streaming to continue, albeit
having lost position tracking in the process. So, query position prior
to seek and use upon failed seek.
Add a boolean multi-channel property with a default of FALSE. When set to TRUE
the element won't mix all input channels to mono, but instead run a FFT on each
channel. In that case the result message would contain a 2 dimensional array
of channel x data for magnitude and phase.
API: GstSpectrum:multi-channel
https://bugzilla.gnome.org/show_bug.cgi?id=593482
Use a separate function to read a sample frame into a ringbuffer slot. In the
future we can use format-specific function pointer to avoid the reoccuring
format checks.
