* ext/srtp/gstsrtp.[ch]: added GST_SRTP_CIPHER_AES_256_ICM to
GstSrtpCipherType and new function cipher_key_size.
* ext/srtp/gstsrtpenc.c: maximum key size is now 46 characters (14 for
the salt plus the key). If different ciphers are chosen for RTP and
RTCP the maximum needed key size is expected.
* ext/srtp/gstsrtpdec.c: minor documentation updates.
https://bugzilla.gnome.org/show_bug.cgi?id=720434
Alternates between 33 and 32 byte frames, but must start
with a 33 byte frame. This has been broken for ages since
the element was ported to the audio decoder base class.
https://bugzilla.gnome.org/show_bug.cgi?id=709416
This currently converts from ARGB64_F16 (16 bit float per component)
to ARGB64 by clipping. We should add support for the F16 format and
implement a conversion filter element that can apply gamma curves,
change exposure, etc.
It only gets the sink flag set when it adds the multifilesink, that
happens in null->ready and it might be too late. Set the flag
explicitly on the constructor.
https://bugzilla.gnome.org/show_bug.cgi?id=711086
This patch fixes three memory leaks in hlsdemux, one that occurs
during normal operation and two that occur during error conditions.
The gst_hls_demux_get_next_fragment function calls
gst_fragment_get_buffer which increments the reference count
on the buffer but gst_hls_demux_get_next_fragment never calls unref on
the buffer. This means that the reference count for each downloaded
fragment never gets to zero and so its memory is never released.
This patch adds a call to gst_buffer_unref after the flags have been
updated on the buffer.
There is a leak-on-error in gst_hls_demux_decrypt_fragment if it fails
to download the key file. If the key fails to download, null is
returned without doing an unref on the encrypted fragment. The
semantics of gst_hls_demux_decrypt_fragment is that it takes ownership
of the encrypted fragment and releases it before returning.
There is a leak-on-error in gst_hls_src_buf_to_utf8_playlist in the
unlikely event that the gst_buffer_map fails. In the "happy path"
operation of gst_hls_src_buf_to_utf8_playlist the buffer gets an unref
before the function returns, therefore the error condition must do the
same.
https://bugzilla.gnome.org/show_bug.cgi?id=710881
Fixed up the error-handling code when downloading fragments.
Modifed the error-handling code to use positive logic when
testing for cancellation of the download loop.
https://bugzilla.gnome.org/show_bug.cgi?id=701404
There is an issue for live streams where download_loop will keep
downloading segments until it gets a 404 error for a segment
that has not yet been published. This is a problem because this
request for a segment that doesn't exist will propagate all the
way back to the origin server(s). This means that dashdemux causes
extra load on the origin server(s) for segments that aren't yet
available.
This patch uses availabilityStartTime, period
and the host's idea of UTC to decide if a fragment is available to
be requested from an HTTP server and filter out requests for fragments
that are not yet available.
https://bugzilla.gnome.org/show_bug.cgi?id=701404
On some live HLS streams, gst_hls_demux_switch_playlist causes
assertion failures because it tried to dereference a NULL fragment.
This is because g_queue_peek_tail sometimes was returning NULL and
this case was not being checked.
This patch does two things:
* move the g_queue_peek_tail inside the semaphore protection
* check if q_queue_peek_tail returns NULL
https://bugzilla.gnome.org/show_bug.cgi?id=708849
gstdashdemux.c:1753: warning: format '%llu' expects type 'long long unsigned int', but argument 8 has type 'long unsigned int'
gstdashdemux.c:2224: warning: format '%llu' expects type 'long long unsigned int', but argument 9 has type 'guint64'
gstdashdemux.c:2224: warning: format '%llu' expects type 'long long unsigned int', but argument 10 has type 'guint64'
gstmpdparser.h:530: warning: type qualifiers ignored on function return type
gstmpdparser.c:4177: warning: type qualifiers ignored on function return type
including the following supports and fixes:
* Create DirectFB surfaces from GstBufferPool
* Add NV12 pixel format support
* Don't use the cursor in the exclusive mode
- EnableCusor() can be only used when the administrative mode is set
in DirectFB 1.6.0 and later.
* Support multiple plane rendering for planar color formats
- This accommodates the chroma plane offsets of the framebuffer
in planar formats.
* Invoke SetConfiguration regardless of video mode setting in setcaps()
- SetConfiguration() method should be invoked regardless of
the result of gst_dfbvideosink_get_best_vmode(), since the two are
unrelated.
* Disable DirectFB signal handler
- "--dfb:no-sighandler" option is passed to DirectFBInit().
This prevents DirectFB from trying to kill the process and allows
GStreamer's termination sequence to proceed normally.
https://bugzilla.gnome.org/show_bug.cgi?id=703520
For SegmentTemplate elements containing a startNumber attribute, the
`number' member of GstMediaSegments should be offset by the value of
startNumber; however, this is not currently the case. As a result, the
first URI(s) requested by the download loop will be wrong.
This commit ensures that segment numbers will be offset by startNumber
when one is present in a SegmentTemplate element.
https://bugzilla.gnome.org/show_bug.cgi?id=705661
When using a SegmentTemplate element, the timestamps of the buffers
output by dashdemux are incorrect, causing problems downstream.
The reason is that GstMediaSegment start times are calculated (in
gst_mpdparser_get_chunk_by_index) by multiplying segment index by
segment duration and then scaling the result according the `timebase'
attribute from the MPD. However, the segment duration is already a
GstClockTime (i.e., it has already been scaled according to the timebase
from the MPD and converted to a nanosecond value), so multiplying it by
the segment index will give the correct timestamp without the need for
any further scaling.
https://bugzilla.gnome.org/show_bug.cgi?id=705679
This prevents locking on startup when a stream only has a single buffer
for one of the streams and mssdemux decides to push an EOS event right
after it.
This prevents deadlocks on startup on files that have only a very
large buffer for a stream and the queue is filled and will lock on
the eos event that is pushed after the buffer. As no buffers have yet
been pushed to other streams, the pipeline locks on preroll
Every encrypted fragment will be a multiple of 128 bits, the last byte
contains the number of bytes that were added as padding in the end
and should be removed.
https://bugzilla.gnome.org/show_bug.cgi?id=701673