When early returning in gst_soup_http_src_read_buffer() because the
element is FLUSHING, we need to unmap and unref the buffer which was just created.
https://bugzilla.gnome.org/show_bug.cgi?id=766718
Directly setting audio/x-raw caps leads to problems when the delivered
data blocks do not align properly at sample boundaries (for example, a
data block with 391 bytes). So, instead, set audio/x-unaligned-raw to
let a parser be autoplugged.
https://bugzilla.gnome.org/show_bug.cgi?id=689460
Non-blocking read will return the amount of data available without
blocking to wait for the full requested size.
The downside is that now it souphttpsrc needs to have a waiting
mechanism in case there is no data available yet to avoid busy
looping arond the inputstream.
The previous ones resulted in odd display aspect ratios and were different
from the ones used by e.g. ffmpeg. The new ones now result in display aspect
ratios of 4:3 and 16:9.
https://bugzilla.gnome.org/show_bug.cgi?id=765946
When a frame's duration is too low, calling gst_util_uint64_scale()
to scale its value can result into it being truncated to zero, which
will cause the vpx encoder to return an VPX_CODEC_INVALID_PARAM error
when trying to encode.
To prevent this from happening, we simply ignore the duration when
encoding if it becomes zero after scaling, logging a warning message.
https://bugzilla.gnome.org/show_bug.cgi?id=765391
This reverts commit 0dd46accf6.
With some audiosinks, starting the ringbuffer on the first commit
causes audio glitches at startup by starting to output segments
from the ringbuffer before it has been filled / fully prerolled. This
doesn't usually happen with pulsesink because we map the pulseaudio
ringbuffer directly, but we should keep things consistent with
other sinks with regards to startup latency, plus it gives more
headway to avoid glitching, should the initial 2nd segment take
more than 10ms to generate.
https://bugzilla.gnome.org/show_bug.cgi?id=657076
Remove calls to gst_pad_has_current_caps() which then go on to call
gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
use gst_pad_get_current_caps() and check for NULL.
https://bugzilla.gnome.org/show_bug.cgi?id=759539
We already pass the entire frame to the decoder. If the decoder ask for
more data, don't pass the same data again as this leads to infinit loop.
Instead, simply fail the fill function to signal the problem with that
frame. It will then be skipped properly.
https://bugzilla.gnome.org/show_bug.cgi?id=761670
All code paths for handle_frame() must somehow take ownership of the frame, be
it by actually unreffing, forwarding the frame elsewhere or storing it for
later.
http://bugzilla.gnome.org/show_bug.cgi?id=760666
With the VPX decoders it's not simple to use downstream buffer pool,
because we don't know the image size and alignment when buffers get
allocated. We can though use GstAllocator (for downstream, or the system
allocator) to avoid a copy before pushing if downstream supports
GstVideoMeta. This would still cause a copy for sink that requires
specialized memory and does not have a GstAllocator for that, though
it will greatly improve performance for sink like glimagesink and
cluttersink. To avoid allocating for every buffer, we also use a
internal buffer pool.
https://bugzilla.gnome.org/show_bug.cgi?id=745372
This adds an automatic mode to the threads property of vpxdec in order to
use as many threads as there is CPU on the platform. This brings back
GStreamer VPX decoding performance closer to what is achieved by other
players, including Chromium.
https://bugzilla.gnome.org/show_bug.cgi?id=758195
When jpeg_finish_decompress is called, output state reference is being created.
But if there is any failures in finishing decompress, it jumps to setjmp,
and at that point state was not referenced. Resulting in leak of output state.
Hence adding another setjmp after output state is referenced.
Similarly adding another setjmp to unmap the frame in case error happens before
finish_decompress
https://bugzilla.gnome.org/show_bug.cgi?id=753087
The problem is that the filesrc and souphttpsrc are behaving
differently regarding the calculation of the segment boundaries. The
filesrc is using a non-inclusive boundaries, while the souphttpsrc
uses inclusive. Currently the hlsdemux calculates the boundaries as
inclusive, so for this reason there is no problem with the souphttpsrc,
but there is an issue in the filesrc.
The GstSegment is non-inclusive, so the proposed solution is to use
non-inclusive boundaries in the hlsdemux in order to be consistent.
Make the change in the hlsdemux, will break the souphttpsrc, which
will expect inclusive boundaries, but the hlsdemux will offer
non-inclusive. This change makes sure that the non-inclusive
boundaries are converted to inclusive.
https://bugzilla.gnome.org/show_bug.cgi?id=748316
These allow a failed request to be retried after the given number of seconds
instead of failing the pipeline. Take account of the Retry-After header if
present. Add retries parameter that controls the number of times an HTTP
request will be retried before failing.
https://bugzilla.gnome.org/show_bug.cgi?id=756318
If soup message is not created then the same should not be passed
on, which is resulting in segfault. Hence throwing a warning message
and returning
https://bugzilla.gnome.org/show_bug.cgi?id=755326
We don't necessarily have full control over the input tags, so
it's possible that the ISRC tag contains a longer string than
expected, in which case we'd write over the end of the static-size
13 byte buffer that is FLAC__StreamMetadata_CueSheet_Track::isrc.
Make sure to only copy the ISRC if it's not too long, and make
sure the buffer we write to is always NUL-terminated by using
g_strlcpy().
CID 1324931.
When passing the taglist to gst_audio_decoder_merge_tags, the reference is increased
by audiodecoder and the caller should free the taglist being passed.
https://bugzilla.gnome.org/show_bug.cgi?id=753903
